• Journal of Korea Multimedia Society
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• v.11 no.3
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• pp.341-352
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• 2008

The dedicated line linked up with the internet maintains generally the efficiency of network by using Best Effort Model. But it is the problem that we allocate the higher bandwidth to services which request high bandwidth regardless of service types or purposes. To overcome this weak point, we use the Session QoS System which distinguish between business services such as E-mail, Web, etc. and non-business services. But, a recent session QoS system is not adequate to the environment similar to university, because that has many users who have various requests. In this paper, to satisfy various requirements of many user's in session QoS system which support services of research, education, business, etc., we analyze types of service in the environment of education and research network and propose a necessity of management strategies of session Qos system.

• Proceedings of the Korean Information Science Society Conference
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• 2002.10e
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• pp.598-600
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• 2002

SIP(Session Initiation Protocol) 프로토콜은 인터넷에서 다양한 세션을 위한 시그널링 프로토콜로서 오디오, 비디오 등의 멀티미디어 세션에 주로 사용되고 있다. 또한 망 유지보수 및 관리가 편리하고 다른 시스템과의 확장성 및 유연성이 뛰어나다는 장점을 갖고 있는 반면 기본적으로 Best Effort 서비스만을 지원하는 IP 네트워크에서 QoS(Quality of Service)를 제공하기 때문에 서비스의 품질에 대한 고려가 미약하다. 따라서 본 연구에서는IP 네트워크를 기반으로 하는 SIP에서 QoS를 제공하기 위해 IntServ(Integrated Services)와 DiffServ(Differentiated Services)를 연동한 QoS 메카니즘을 제시한다.

• Proceedings of the Korea Society of Information Technology Applications Conference
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• 2007.05a
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• pp.29-37
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• 2007

장소와 시간에 무관하게 누구와도 하나의 단말기로 멀티미디어 통신을 할 수 있게 하기 위한 이동통신 시스템이 단일표준으로 되지 못하고 복수표준으로 진행됨에 따라 이종의 접속망들로 구성되는 융합망이 불가피하게 되었다. 이러한 융합망 환경에서는 멀티미디어 서비스의 지속성을 유지하기 위한 QoS 보장 방안이 필수적이다. 본 논문에서는 멀티미디어 세션의 지연이나 손실 등을 지속적으로 관찰하고 계약된 SLA 의 QoS 기준 만족 여부를 평가하고 이를 바탕으로 이종 망 내에서의 QoS 적응 및 조정을 수행함으써 멀티미디어 서비스의 QoS 를 유지하는 방안을 제안한다.

• Journal of KIISE:Computer Systems and Theory
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• v.26 no.9
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• pp.1112-1125
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• 1999

최근 들어 멀티미디어를 지원하는 응용들의 서비스 품질 보장과 멀티캐스트 지원에 대한 요구가 증가되고 있으나 기존의 멀티캐스트 프로토콜로는 이러한 요구를 수용할 수 없는 문제가 발생하고 있다. 현재 인터넷에서 사용되고 있는 라우팅 메커니즘은 네트워크 자원 정보나 세션의 QoS 요구사항을 고려하지 않고 단순히 종단간의 연결에만 초점을 맞추고 있기 때문이다. 따라서 본 논문에서는 멀티캐스트 환경에서 서비스 품질보장을 지원하기 위한 새로운 프로토콜, LayeredQoS을 제안한다. 다중의 CP (Central Point)를 채택하고 각각의 CP에 적절한 QoS 레벨을 부여하여 사용함으로써 대역폭의 공유정도를 높일 뿐만 아니라 전체 트리 비용을 감소시켜 궁극적으로 네트워크 처리량이 증가되도록 하였다. 또한 시뮬레이션 방법을 통하여 다른 프로토콜보다 나은 성능을 가지는 것으로 평가하였다.Abstract Many emerging multimedia applications often require a guaranteed quality of service and multicast connection. But the traditional multicast protocol can't meet the needs since the routing mechanisms deployed in today's Internet are focused on connectivity, not on resource availability in the network or QoS requirements of flows. In this paper, we present LayeredQoS, a new QoS-based multicast routing algorithm. We adopt the multiple CPs(Central Points) and allocate QoS-levels for each CP in order to improve the degree of resource sharing and decrease the total tree cost, and then network throughput is increased. The proposed protocol is verified by simulations and it is shown that the performance of LayeredQoS is much better than the existing protocols.

• Journal of KIISE:Computing Practices and Letters
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• v.5 no.2
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• pp.157-167
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• 1999

편리한 사용 방법과 멀티미디어 데이터 서비스를 지원하는 웹은 현 인터넷 환경에서 이종 컴퓨터간의 주요한 정보 전달 수단으로 자리잡았다. 그러나 웹이 사용하고 있는 통신 프로토콜 스택만으로는 부상하고 있는 정보 형태인 실시간 멀티미디어 데이터의 서비스를 적절히 지원할수 없었다. 이에 본 논문에서는 실시간 전송 프로토콜인 RTP/RTCP를 웹 통신 프로토콜 스택에 추가하여 실시간 멀티미디어 세션의 서비스를 지원할수 잇도록 그 기능이 확장된 웹 시스템의 구현을 소개한다. 이는 MBone을 통해 실시간으로 멀티캐스트되고 있는 멀티캐스트 되고 있는 멀티미디어 세션의 서비스를 웹에서 지원할수 있도록 한 것으로, 기존의 MBone응용들을 수용한 형태를 가진다. 이를 위하여, MBone상에서 개설된 세션 정보를 수집하여 이를 관리하고 그 정보들을 웹 문서로 제공하는 프로그램인 '웹 SDR'과 MBone을 통해 멀티캐스트되는 실시간 멀티미디어 세션을 웹상에서 시청할수 있도록 웹 브라우저의 기능을 확장한 Windows 95 플랫폼 기반의 '실시간 멀티미디어 웹 플러그인'을 구현하였다 그리고 , 본 연구에서 개발한 수신 위주의 웹 시스템의 서비스 품질을 개선하기 위하여 현 인터넷 환경을 고려한 서비스 품질 제어 기법인 '관측 기반의 수신단 QoS 제어 알고리즘'을 제안하고, 이를 적용하여 실시간세션 서비스 품질이 개선됨을 보였다.

• The Journal of Korea Institute of Information, Electronics, and Communication Technology
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• v.8 no.1
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• pp.18-22
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• 2015

The relationship of information collection and utility is relationship of man to man, but is developed object to object by information communication and control system. This paper suggests a multimedia collaboration environment based on M2M(Machine to machine). We propose a study on reliability for application QoS in distributed multimedia. When it recovers, the characteristic of this system is to use the same method to get back again it as it creates a session.

• Convergence Security Journal
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• v.8 no.4
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• pp.41-49
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• 2008

VoIP is a service that changes the analogue audio signal into a digital signal and then transfers the audio information to the users after configuring it as a packet; and it has an advantage of lower price than the existing voice call service and better extensibility. However, VoIP service has a system structure that, compared to the existing PSTN (Public Switched Telephone Network), has poor call quality and is vulnerable in the security aspect. To make up these problems, TLS service was introduced to enhance the security. In practical system, however, since QoS problem occurs, it is necessary to develop the VoIP security system that can satisfy QoS at the same time in the security aspect. In this paper, a user authentication VoIP system that can provide a service according to the security and the user through providing a differential service according to the approach of the users by adding AA server at the step of configuring the existing VoIP session is suggested. It was found that the proposed system of this study provides a quicker QoS than the TLS-added system at a similar level of security. Also, it is able to provide a variety of additional services by the different users.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.35 no.8A
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• pp.820-827
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• 2010

In these days, video applications for special purposes such as video conference systems among multiple users and video surveillance systems require multiple video connections and QoS guarantee. The video systems employ IEEE 802.11 Wireless LAN devices to support broadband wireless interfaces and easy internet accesses for cheaper prices. However, according to the current IEEE 802.11e HCCA standard, if more than three video sessions are established in WSTA services, some of them must share the TXOP because the available number of TSIDs for video transmission is two. In order to resolve the problem, we devised a method which can establish up to 15 video sessions by slightly modifying the frame structure while maintaining the compatibility with current standard. Our method is implemented on the NCTUns 4.0 network simulator, and evaluated not only numerically in terms of throughput, delay, and PSNR, but also experimentally in the sense of real video clips that are used as input to our simulation. The results showed that our method sufficiently guarantees the transmission bandwidth requested by each video session.

• The KIPS Transactions:PartC
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• v.14C no.5
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• pp.453-462
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• 2007

NGN is a packet-based converged network to support session and non-session based services in QoS-enabled broadband transport technologies. Resource and admission control functions(RACF) of NGN have been considering COPS and SNMP as resource management protocols to collect network topology and resource status information of transport network. This paper defines requirements of resource management in NGN transport network, and proposes protocol independent information model to keep consistency of data models of each resource management protocol. Based on this information model described with UML class diagram, we designed and implemented NGN resource management system on Web Services.

• Journal of Korea Multimedia Society
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• v.3 no.5
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• pp.479-494
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• 2000

Although the proliferation of real-time multimedia services over the Internet might indicate its successfulness in dealing with heterogeneous environments, it is obvious, on the other hand, that the internet now has to cope with a flood of multimedia data which consumes most of network communication channels due to a great deal of video or audio streams. Therefore, for the purpose of an efficient and appropriate utilization of network resources, it requires to develop and deploy a new scalable transmission technique n consideration of respective network environment and individual clients computing power. Also, we can eliminate the waste effects of storage device and data transmission overhead in that the same video stream duplicated according to QoS. The purpose of this paper is to develop a technology that can adjust the amount of data transmitted as an MPEG video stream according to its given communication bandwidth, and technique that can reflect dynamic bandwidth while playing a video stream. For this purpose, we introduce a media scalable media decomposer working on server side, and a scalable media composer working o n a client side, and then propose a scalable transmission method and a media sender and a media receiver in consideration of dynamic QoS. Those methods proposed her can facilitate an effective use of network resources, and provide multimedia MPEG video services in real-time with respect to individual client computing environment.